Bug 1912989 - Vendor libwebrtc from aefed55c25

Upstream commit: https://webrtc.googlesource.com/src/+/aefed55c2529310235d4414074a74eed17b958da [iwyu][1\n] Applying to api/[a-s]* First batch of applying iwyu to the repo. Done with: > ./tools_webrtc/iwyu/apply-iwyu api > git add api/[a-s]* > python3 gn_autodeps.py ~/local/webrtc/src out/Default Last step is a custom script I wrote to automatically apply new required dependencies for target in gn, which saved tons of time manually going over the files and fixing. If this is something that interest others, I can submit it as well. Bug: webrtc:42226242 Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Dor Hen <dorhen@meta.com> Cr-Commit-Position: refs/heads/main@{#42512}
2024-11-24 05:11:16 +00:00 · 2024-08-26 16:02:42 -06:00 · 2024-08-26 16:02:42 -06:00 · 9dc59000b6
commit 9dc59000b6
parent ccf1ef88a1
100 changed files with 424 additions and 105 deletions
--- a/third_party/libwebrtc/README.moz-ff-commit
+++ b/third_party/libwebrtc/README.moz-ff-commit
@ -31311,3 +31311,6 @@ d4a6c3f76f
 # MOZ_LIBWEBRTC_SRC=/Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
 # base of lastest vendoring
 e7a305d3c7
+# MOZ_LIBWEBRTC_SRC=/Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc MOZ_LIBWEBRTC_BRANCH=mozpatches bash dom/media/webrtc/third_party_build/fast-forward-libwebrtc.sh
+# base of lastest vendoring
+aefed55c25
--- a/third_party/libwebrtc/README.mozilla
+++ b/third_party/libwebrtc/README.mozilla
@ -20898,3 +20898,5 @@ libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc co
 libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-08-26T21:57:34.717308.
 # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
 libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-08-26T22:00:12.468466.
+# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2024-08-26T22:02:22.879432.
--- a/third_party/libwebrtc/api/BUILD.gn
+++ b/third_party/libwebrtc/api/BUILD.gn
@ -34,7 +34,11 @@ rtc_source_set("enable_media") {
  ]
  deps = [
    ":libjingle_peerconnection_api",
+    ":scoped_refptr",
    "../call",
+    "../call:call_interfaces",
+    "../call:call_interfaces",
+    "../media:media_engine",
    "../media:rtc_audio_video",
    "../pc:media_factory",
    "../rtc_base/system:rtc_export",
@ -59,13 +63,19 @@ rtc_source_set("enable_media_with_defaults") {
  deps = [
    ":enable_media",
    ":libjingle_peerconnection_api",
+    ":scoped_refptr",
    "../rtc_base/system:rtc_export",
    "audio:audio_processing",
+    "audio_codecs:audio_codecs_api",
+    "audio_codecs:audio_codecs_api",
    "audio_codecs:builtin_audio_decoder_factory",
    "audio_codecs:builtin_audio_encoder_factory",
    "task_queue:default_task_queue_factory",
+    "task_queue:task_queue",
    "video_codecs:builtin_video_decoder_factory",
    "video_codecs:builtin_video_encoder_factory",
+    "video_codecs:video_codecs_api",
+    "video_codecs:video_codecs_api",
  ]
 }

@ -85,6 +95,7 @@ if (!build_with_chromium && !build_with_mozilla) {
      "../api/rtc_event_log:rtc_event_log_factory",
      "../pc:peer_connection_factory",
      "../pc:webrtc_sdp",
+      "../rtc_base:socket_server",
      "../rtc_base:threading",
      "../rtc_base/system:rtc_export",
      "../stats:rtc_stats",
@ -116,6 +127,8 @@ rtc_library("rtp_headers") {
  ]
  deps = [
    ":array_view",
+    "../rtc_base:checks",
+    "../rtc_base/system:rtc_export",
    "units:timestamp",
    "video:video_rtp_headers",
    "//third_party/abseil-cpp/absl/types:optional",
@ -237,6 +250,7 @@ if (!build_with_mozilla) {
    ":scoped_refptr",
    "../rtc_base:ssl",
    "../rtc_base/system:rtc_export",
+    "//third_party/abseil-cpp/absl/base:core_headers",
    "//third_party/abseil-cpp/absl/types:optional",
  ]
 }
@ -366,14 +380,19 @@ if (!build_with_mozilla) {
    "../pc:media_factory",
    "../rtc_base:copy_on_write_buffer",
    "../rtc_base:logging",
+    "../rtc_base:macromagic",
    "../rtc_base:network",
    "../rtc_base:network_constants",
    "../rtc_base:rtc_certificate_generator",
+    "../rtc_base:socket_factory",
    "../rtc_base:ssl",
    "../rtc_base:ssl_adapter",
    "../rtc_base:stringutils",
    "adaptation:resource_adaptation_api",
+    "audio:audio_device",
+    "audio:audio_frame_processor",
    "audio:audio_mixer_api",
+    "audio:audio_processing",
    "audio_codecs:audio_codecs_api",
    "crypto:frame_decryptor_interface",
    "crypto:frame_encryptor_interface",
@ -389,6 +408,7 @@ if (!build_with_mozilla) {
    "transport:sctp_transport_factory_interface",
    "transport/rtp:rtp_source",
    "units:data_rate",
+    "units:time_delta",
    "units:timestamp",
    "video:encoded_image",
    "video:video_bitrate_allocator_factory",
@ -971,12 +991,14 @@ rtc_library("ice_transport_factory") {
    ":make_ref_counted",
    ":packet_socket_factory",
    ":scoped_refptr",
+    ":sequence_checker",
    "../p2p:connection",
    "../p2p:ice_transport_internal",
    "../p2p:p2p_constants",
    "../p2p:p2p_transport_channel",
    "../p2p:port_allocator",
    "../p2p:rtc_p2p",
+    "../rtc_base:macromagic",
    "../rtc_base:threading",
    "../rtc_base/system:rtc_export",
    "rtc_event_log:rtc_event_log",
@ -1006,6 +1028,7 @@ rtc_source_set("sequence_checker") {
    "../rtc_base:checks",
    "../rtc_base:macromagic",
    "../rtc_base/synchronization:sequence_checker_internal",
+    "task_queue:task_queue",
  ]
 }

@ -1489,6 +1512,8 @@ if (rtc_include_tests) {
      ":peer_connection_quality_test_fixture_api",
      ":rtc_error",
      ":rtc_event_log_output_file",
+      ":rtp_headers",
+      ":rtp_headers",
      ":rtp_packet_info",
      ":rtp_parameters",
      ":scoped_refptr",
@ -1498,11 +1523,14 @@ if (rtc_include_tests) {
      "../rtc_base:buffer",
      "../rtc_base:checks",
      "../rtc_base:gunit_helpers",
+      "../rtc_base:macromagic",
      "../rtc_base:platform_thread",
      "../rtc_base:rtc_event",
+      "../rtc_base:socket_address",
      "../rtc_base:ssl",
      "../rtc_base:task_queue_for_test",
      "../rtc_base/containers:flat_set",
+      "../rtc_base/synchronization:sequence_checker_internal",
      "../rtc_base/task_utils:repeating_task",
      "../system_wrappers:field_trial",
      "../test:field_trial",
@ -1520,6 +1548,9 @@ if (rtc_include_tests) {
      "video:frame_buffer_unittest",
      "video:rtp_video_frame_assembler_unittests",
      "video:video_frame_metadata_unittest",
+      "//testing/gtest",
+      "//third_party/abseil-cpp/absl/functional:any_invocable",
+      "//third_party/abseil-cpp/absl/strings",
      "//third_party/abseil-cpp/absl/strings:string_view",
      "//third_party/abseil-cpp/absl/types:optional",
    ]
@ -1622,6 +1653,8 @@ rtc_library("frame_transformer_factory") {
    ":scoped_refptr",
    "../audio:audio",
    "../modules/rtp_rtcp",
+    "../rtc_base:checks",
+    "../rtc_base/system:rtc_export",
    "video:encoded_frame",
    "video:video_frame_metadata",
  ]
--- a/third_party/libwebrtc/api/DEPS
+++ b/third_party/libwebrtc/api/DEPS
@ -116,6 +116,7 @@ specific_include_rules = {
    "+rtc_base/rtc_certificate.h",
    "+rtc_base/rtc_certificate_generator.h",
    "+rtc_base/socket_address.h",
+    "+rtc_base/socket_factory.h",
    "+rtc_base/ssl_certificate.h",
    "+rtc_base/ssl_stream_adapter.h",
    "+rtc_base/thread.h",
@ -140,6 +141,7 @@ specific_include_rules = {
  ],

  "legacy_stats_types\.h": [
+    "+rtc_base/thread_annotations.h",
    "+rtc_base/thread_checker.h",
  ],

--- a/third_party/libwebrtc/api/array_view.h
+++ b/third_party/libwebrtc/api/array_view.h
@ -13,6 +13,7 @@

 #include <algorithm>
 #include <array>
+#include <cstddef>
 #include <iterator>
 #include <type_traits>

--- a/third_party/libwebrtc/api/array_view_unittest.cc
+++ b/third_party/libwebrtc/api/array_view_unittest.cc
@ -10,16 +10,18 @@

 #include "api/array_view.h"

-#include <algorithm>
+#include <stdint.h>
+
 #include <array>
+#include <cstddef>
 #include <string>
 #include <utility>
 #include <vector>

 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/gunit.h"
 #include "test/gmock.h"
+#include "test/gtest.h"

 namespace rtc {

--- a/third_party/libwebrtc/api/audio/BUILD.gn
+++ b/third_party/libwebrtc/api/audio/BUILD.gn
@ -41,6 +41,7 @@ rtc_library("audio_frame_api") {
    "../../rtc_base:logging",
    "../../rtc_base:macromagic",
    "../../rtc_base:timeutils",
+    "//third_party/abseil-cpp/absl/types:optional",
  ]
 }

@ -124,6 +125,7 @@ rtc_library("aec3_factory") {
    ":echo_control",
    "../../modules/audio_processing/aec3",
    "../../rtc_base/system:rtc_export",
+    "//third_party/abseil-cpp/absl/types:optional",
  ]
 }

--- a/third_party/libwebrtc/api/audio/audio_frame.cc
+++ b/third_party/libwebrtc/api/audio/audio_frame.cc
@ -12,6 +12,13 @@

 #include <string.h>

+#include <cstdint>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/audio/audio_view.h"
+#include "api/audio/channel_layout.h"
+#include "api/rtp_packet_infos.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/time_utils.h"

--- a/third_party/libwebrtc/api/audio/audio_frame.h
+++ b/third_party/libwebrtc/api/audio/audio_frame.h
@ -16,6 +16,7 @@

 #include <array>

+#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/audio/audio_view.h"
 #include "api/audio/channel_layout.h"
--- a/third_party/libwebrtc/api/audio/echo_canceller3_factory.cc
+++ b/third_party/libwebrtc/api/audio/echo_canceller3_factory.cc
@ -11,6 +11,9 @@

 #include <memory>

+#include "absl/types/optional.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_control.h"
 #include "modules/audio_processing/aec3/echo_canceller3.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/audio_codecs/BUILD.gn
+++ b/third_party/libwebrtc/api/audio_codecs/BUILD.gn
@ -46,6 +46,7 @@ rtc_library("audio_codecs_api") {
    "../units:data_rate",
    "../units:time_delta",
    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/base:nullability",
    "//third_party/abseil-cpp/absl/strings",
    "//third_party/abseil-cpp/absl/strings:string_view",
    "//third_party/abseil-cpp/absl/types:optional",
@ -103,13 +104,18 @@ rtc_library("builtin_audio_encoder_factory") {
  ]
  defines = []
  if (rtc_include_ilbc) {
-    deps += [ "ilbc:audio_encoder_ilbc" ]
+    deps += [
+      "..:field_trials_view",
+      "ilbc:audio_encoder_ilbc",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
    defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
  } else {
    defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
  }
  if (rtc_include_opus) {
    deps += [
+      "..:field_trials_view",
      "opus:audio_encoder_multiopus",
      "opus:audio_encoder_opus",
    ]
--- a/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn
+++ b/third_party/libwebrtc/api/audio_codecs/L16/BUILD.gn
@ -23,6 +23,7 @@ rtc_library("audio_encoder_L16") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:pcm16b",
+    "../../../rtc_base:checks",
    "../../../rtc_base:safe_conversions",
    "../../../rtc_base:safe_minmax",
    "../../../rtc_base:stringutils",
--- a/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.cc
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_decoder_L16.cc
@ -11,8 +11,14 @@
 #include "api/audio_codecs/L16/audio_decoder_L16.h"

 #include <memory>
+#include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
 #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
 #include "rtc_base/numerics/safe_conversions.h"
--- a/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc
+++ b/third_party/libwebrtc/api/audio_codecs/L16/audio_encoder_L16.cc
@ -10,11 +10,22 @@

 #include "api/audio_codecs/L16/audio_encoder_L16.h"

+#include <stddef.h>
+
+#include <map>
 #include <memory>
+#include <utility>
+#include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
 #include "rtc_base/string_to_number.h"
--- a/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h
+++ b/third_party/libwebrtc/api/audio_codecs/audio_decoder_factory_template.h
@ -12,9 +12,14 @@
 #define API_AUDIO_CODECS_AUDIO_DECODER_FACTORY_TEMPLATE_H_

 #include <memory>
+#include <type_traits>
 #include <vector>

+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
 #include "api/field_trials_view.h"
 #include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
--- a/third_party/libwebrtc/api/audio_codecs/audio_encoder.h
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder.h
@ -11,6 +11,9 @@
 #ifndef API_AUDIO_CODECS_AUDIO_ENCODER_H_
 #define API_AUDIO_CODECS_AUDIO_ENCODER_H_

+#include <stddef.h>
+#include <stdint.h>
+
 #include <memory>
 #include <string>
 #include <utility>
--- a/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h
+++ b/third_party/libwebrtc/api/audio_codecs/audio_encoder_factory_template.h
@ -12,9 +12,15 @@
 #define API_AUDIO_CODECS_AUDIO_ENCODER_FACTORY_TEMPLATE_H_

 #include <memory>
+#include <type_traits>
 #include <vector>

+#include "absl/base/nullability.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
 #include "api/environment/environment.h"
 #include "api/field_trials_view.h"
 #include "api/make_ref_counted.h"
--- a/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn
+++ b/third_party/libwebrtc/api/audio_codecs/g711/BUILD.gn
@ -23,6 +23,7 @@ rtc_library("audio_encoder_g711") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:checks",
    "../../../rtc_base:safe_conversions",
    "../../../rtc_base:safe_minmax",
    "../../../rtc_base:stringutils",
@ -43,6 +44,7 @@ rtc_library("audio_decoder_g711") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:g711",
+    "../../../rtc_base:checks",
    "../../../rtc_base:safe_conversions",
    "../../../rtc_base/system:rtc_export",
    "//third_party/abseil-cpp/absl/strings",
--- a/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_decoder_g711.cc
@ -10,11 +10,18 @@

 #include "api/audio_codecs/g711/audio_decoder_g711.h"

+#include <initializer_list>
 #include <memory>
 #include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc
+++ b/third_party/libwebrtc/api/audio_codecs/g711/audio_encoder_g711.cc
@ -10,11 +10,22 @@

 #include "api/audio_codecs/g711/audio_encoder_g711.h"

+#include <stddef.h>
+
+#include <initializer_list>
+#include <map>
 #include <memory>
+#include <utility>
 #include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
 #include "rtc_base/string_to_number.h"
--- a/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn
+++ b/third_party/libwebrtc/api/audio_codecs/g722/BUILD.gn
@ -30,6 +30,7 @@ rtc_library("audio_encoder_g722") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:g722",
+    "../../../rtc_base:checks",
    "../../../rtc_base:safe_conversions",
    "../../../rtc_base:safe_minmax",
    "../../../rtc_base:stringutils",
@ -50,6 +51,7 @@ rtc_library("audio_decoder_g722") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:g722",
+    "../../../rtc_base:checks",
    "../../../rtc_base:safe_conversions",
    "../../../rtc_base/system:rtc_export",
    "//third_party/abseil-cpp/absl/strings",
--- a/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_decoder_g722.cc
@ -14,7 +14,13 @@
 #include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc
+++ b/third_party/libwebrtc/api/audio_codecs/g722/audio_encoder_g722.cc
@ -10,11 +10,22 @@

 #include "api/audio_codecs/g722/audio_encoder_g722.h"

+#include <stddef.h>
+
+#include <map>
 #include <memory>
+#include <utility>
 #include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
 #include "rtc_base/string_to_number.h"
--- a/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/BUILD.gn
@ -29,6 +29,7 @@ rtc_library("audio_encoder_ilbc") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:ilbc",
+    "../../../rtc_base:checks",
    "../../../rtc_base:safe_conversions",
    "../../../rtc_base:safe_minmax",
    "../../../rtc_base:stringutils",
--- a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
@ -14,6 +14,11 @@
 #include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
+++ b/third_party/libwebrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
@ -10,12 +10,21 @@

 #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"

+#include <map>
 #include <memory>
+#include <string>
+#include <utility>
 #include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
-#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
 #include "rtc_base/string_to_number.h"

--- a/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn
+++ b/third_party/libwebrtc/api/audio_codecs/opus/BUILD.gn
@ -48,6 +48,7 @@ rtc_library("audio_encoder_opus") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base:checks",
    "../../../rtc_base/system:rtc_export",
    "//third_party/abseil-cpp/absl/types:optional",
  ]
@ -64,6 +65,7 @@ rtc_library("audio_decoder_opus") {
    "..:audio_codecs_api",
    "../../../api:field_trials_view",
    "../../../modules/audio_coding:webrtc_opus",
+    "../../../rtc_base:checks",
    "../../../rtc_base/system:rtc_export",
    "//third_party/abseil-cpp/absl/strings",
    "//third_party/abseil-cpp/absl/types:optional",
--- a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_multi_channel_opus.cc
@ -14,8 +14,12 @@
 #include <utility>
 #include <vector>

-#include "absl/memory/memory.h"
-#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/opus/audio_decoder_multi_channel_opus_impl.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_decoder_opus.cc
@ -10,12 +10,20 @@

 #include "api/audio_codecs/opus/audio_decoder_opus.h"

+#include <map>
 #include <memory>
+#include <string>
 #include <utility>
 #include <vector>

 #include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "rtc_base/checks.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h
@ -12,11 +12,8 @@
 #define API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_CONFIG_H_

 #include <stddef.h>
-
 #include <vector>

-#include "absl/types/optional.h"
-#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
 #include "rtc_base/system/rtc_export.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc
+++ b/third_party/libwebrtc/api/audio_codecs/opus/audio_encoder_opus.cc
@ -10,7 +10,17 @@

 #include "api/audio_codecs/opus/audio_encoder_opus.h"

+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "api/field_trials_view.h"
 #include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "rtc_base/checks.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn
+++ b/third_party/libwebrtc/api/audio_codecs/test/BUILD.gn
@ -21,6 +21,8 @@ if (rtc_include_tests) {
    ]
    deps = [
      "..:audio_codecs_api",
+      "../..:make_ref_counted",
+      "../..:scoped_refptr",
      "../../../test:audio_codec_mocks",
      "../../../test:scoped_key_value_config",
      "../../../test:test_support",
@ -36,6 +38,7 @@ if (rtc_include_tests) {
      "../ilbc:audio_encoder_ilbc",
      "../opus:audio_decoder_opus",
      "../opus:audio_encoder_opus",
+      "//third_party/abseil-cpp/absl/types:optional",
    ]
  }
 }
--- a/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
+++ b/third_party/libwebrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
@ -11,12 +11,21 @@
 #include "api/audio_codecs/audio_decoder_factory_template.h"

 #include <memory>
+#include <utility>
+#include <vector>

+#include "absl/types/optional.h"
 #include "api/audio_codecs/L16/audio_decoder_L16.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/g711/audio_decoder_g711.h"
 #include "api/audio_codecs/g722/audio_decoder_g722.h"
 #include "api/audio_codecs/ilbc/audio_decoder_ilbc.h"
 #include "api/audio_codecs/opus/audio_decoder_opus.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_audio_decoder.h"
--- a/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
+++ b/third_party/libwebrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
@ -11,14 +11,23 @@
 #include "api/audio_codecs/audio_encoder_factory_template.h"

 #include <memory>
+#include <utility>
+#include <vector>

+#include "absl/types/optional.h"
 #include "api/audio_codecs/L16/audio_encoder_L16.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/g711/audio_encoder_g711.h"
 #include "api/audio_codecs/g722/audio_encoder_g722.h"
 #include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
 #include "api/audio_codecs/opus/audio_encoder_opus.h"
 #include "api/environment/environment.h"
 #include "api/environment/environment_factory.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_audio_encoder.h"
--- a/third_party/libwebrtc/api/audio_options.h
+++ b/third_party/libwebrtc/api/audio_options.h
@ -11,8 +11,6 @@
 #ifndef API_AUDIO_OPTIONS_H_
 #define API_AUDIO_OPTIONS_H_

-#include <stdint.h>
-
 #include <string>

 #include "absl/types/optional.h"
--- a/third_party/libwebrtc/api/call/transport.cc
+++ b/third_party/libwebrtc/api/call/transport.cc
@ -10,8 +10,6 @@

 #include "api/call/transport.h"

-#include <cstdint>
-
 namespace webrtc {

 PacketOptions::PacketOptions() = default;
--- a/third_party/libwebrtc/api/call/transport.h
+++ b/third_party/libwebrtc/api/call/transport.h
@ -11,7 +11,6 @@
 #ifndef API_CALL_TRANSPORT_H_
 #define API_CALL_TRANSPORT_H_

-#include <stddef.h>
 #include <stdint.h>

 #include "api/array_view.h"
--- a/third_party/libwebrtc/api/candidate.cc
+++ b/third_party/libwebrtc/api/candidate.cc
@ -10,12 +10,19 @@

 #include "api/candidate.h"

-#include "absl/base/attributes.h"
+#include <algorithm>  // IWYU pragma: keep
+#include <cstdint>
+#include <string>
+
+#include "absl/strings/string_view.h"
 #include "p2p/base/p2p_constants.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/crc32.h"
 #include "rtc_base/crypto_random.h"
 #include "rtc_base/ip_address.h"
-#include "rtc_base/logging.h"
+#include "rtc_base/network_constants.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/string_encode.h"
 #include "rtc_base/strings/string_builder.h"

 using webrtc::IceCandidateType;
--- a/third_party/libwebrtc/api/candidate.h
+++ b/third_party/libwebrtc/api/candidate.h
@ -11,13 +11,11 @@
 #ifndef API_CANDIDATE_H_
 #define API_CANDIDATE_H_

-#include <limits.h>
+#include <stddef.h>
 #include <stdint.h>

-#include <algorithm>
 #include <string>

-#include "absl/base/attributes.h"
 #include "absl/strings/string_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/network_constants.h"
--- a/third_party/libwebrtc/api/candidate_unittest.cc
+++ b/third_party/libwebrtc/api/candidate_unittest.cc
@ -13,7 +13,8 @@
 #include <string>

 #include "p2p/base/p2p_constants.h"
-#include "rtc_base/gunit.h"
+#include "rtc_base/socket_address.h"
+#include "test/gtest.h"

 using webrtc::IceCandidateType;

--- a/third_party/libwebrtc/api/create_peerconnection_factory.h
+++ b/third_party/libwebrtc/api/create_peerconnection_factory.h
@ -10,6 +10,7 @@

 #ifndef API_CREATE_PEERCONNECTION_FACTORY_H_
 #define API_CREATE_PEERCONNECTION_FACTORY_H_
+// IWYU pragma: no_include "rtc_base/thread.h"

 #include <memory>

@ -29,11 +30,9 @@ namespace rtc {
 // TODO(bugs.webrtc.org/9987): Move rtc::Thread to api/ or expose a better
 // type. At the moment, rtc::Thread is not part of api/ so it cannot be
 // included in order to avoid to leak internal types.
-class Thread;
+class Thread;  // IWYU pragma: keep
 }  // namespace rtc
-
 namespace webrtc {
-
 class AudioFrameProcessor;

 // Create a new instance of PeerConnectionFactoryInterface with optional video
--- a/third_party/libwebrtc/api/crypto/BUILD.gn
+++ b/third_party/libwebrtc/api/crypto/BUILD.gn
@ -23,6 +23,7 @@ rtc_library("options") {
    "crypto_options.h",
  ]
  deps = [
+    "../../rtc_base:checks",
    "../../rtc_base:ssl_adapter",
    "../../rtc_base/system:rtc_export",
  ]
--- a/third_party/libwebrtc/api/crypto/crypto_options.cc
+++ b/third_party/libwebrtc/api/crypto/crypto_options.cc
@ -10,6 +10,9 @@

 #include "api/crypto/crypto_options.h"

+#include <vector>
+
+#include "rtc_base/checks.h"
 #include "rtc_base/ssl_stream_adapter.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/dtls_transport_interface.cc
+++ b/third_party/libwebrtc/api/dtls_transport_interface.cc
@ -10,6 +10,12 @@

 #include "api/dtls_transport_interface.h"

+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "rtc_base/ssl_certificate.h"
+
 namespace webrtc {

 DtlsTransportInformation::DtlsTransportInformation()
--- a/third_party/libwebrtc/api/dtls_transport_interface.h
+++ b/third_party/libwebrtc/api/dtls_transport_interface.h
@ -12,8 +12,8 @@
 #define API_DTLS_TRANSPORT_INTERFACE_H_

 #include <memory>
-#include <utility>

+#include "absl/base/attributes.h"
 #include "absl/types/optional.h"
 #include "api/ice_transport_interface.h"
 #include "api/ref_count.h"
--- a/third_party/libwebrtc/api/enable_media.cc
+++ b/third_party/libwebrtc/api/enable_media.cc
@ -15,8 +15,11 @@

 #include "api/environment/environment.h"
 #include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "call/call.h"
+#include "call/call_config.h"
 #include "call/create_call.h"
-#include "media/engine/webrtc_media_engine.h"
+#include "media/base/media_engine.h"
 #include "media/engine/webrtc_video_engine.h"
 #include "media/engine/webrtc_voice_engine.h"
 #include "pc/media_factory.h"
--- a/third_party/libwebrtc/api/enable_media_with_defaults.cc
+++ b/third_party/libwebrtc/api/enable_media_with_defaults.cc
@ -10,10 +10,14 @@

 #include "api/enable_media_with_defaults.h"

+#include <memory>
+
 #include "api/audio/audio_processing.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/enable_media.h"
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/video_codecs/builtin_video_decoder_factory.h"
 #include "api/video_codecs/builtin_video_encoder_factory.h"
--- a/third_party/libwebrtc/api/environment/BUILD.gn
+++ b/third_party/libwebrtc/api/environment/BUILD.gn
@ -28,6 +28,7 @@ rtc_library("environment_factory") {
  ]
  deps = [
    ":environment",
+    "..:field_trials_view",
    "..:make_ref_counted",
    "..:refcountedbase",
    "..:scoped_refptr",
@ -36,6 +37,7 @@ rtc_library("environment_factory") {
    "../../system_wrappers",
    "../rtc_event_log",
    "../task_queue:default_task_queue_factory",
+    "../task_queue:task_queue",
    "../transport:field_trial_based_config",
    "//third_party/abseil-cpp/absl/base:nullability",
  ]
@ -55,6 +57,7 @@ if (rtc_include_tests) {
      "../task_queue",
      "../units:timestamp",
      "//third_party/abseil-cpp/absl/functional:any_invocable",
+      "//third_party/abseil-cpp/absl/strings:string_view",
      "//third_party/abseil-cpp/absl/types:optional",
    ]
  }
--- a/third_party/libwebrtc/api/environment/environment_factory.cc
+++ b/third_party/libwebrtc/api/environment/environment_factory.cc
@ -12,11 +12,16 @@

 #include <memory>
 #include <utility>
-#include <vector>

+#include "absl/base/nullability.h"
+#include "api/environment/environment.h"
+#include "api/field_trials_view.h"
 #include "api/make_ref_counted.h"
+#include "api/ref_counted_base.h"
 #include "api/rtc_event_log/rtc_event_log.h"
+#include "api/scoped_refptr.h"
 #include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_factory.h"
 #include "api/transport/field_trial_based_config.h"
 #include "rtc_base/checks.h"
 #include "system_wrappers/include/clock.h"
--- a/third_party/libwebrtc/api/environment/environment_factory.h
+++ b/third_party/libwebrtc/api/environment/environment_factory.h
@ -24,11 +24,12 @@ namespace webrtc {

 // These classes are forward declared to reduce amount of headers exposed
 // through api header.
+// IWYU pragma: begin_keep
 class Clock;
 class TaskQueueFactory;
 class FieldTrialsView;
 class RtcEventLog;
-
+// IWYU pragma: end_keep
 // Constructs `Environment`.
 // Individual utilities are provided using one of the `Set` functions.
 // `Set` functions do nothing when nullptr value is passed.
--- a/third_party/libwebrtc/api/environment/environment_unittest.cc
+++ b/third_party/libwebrtc/api/environment/environment_unittest.cc
@ -16,10 +16,13 @@
 #include <vector>

 #include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/environment/environment_factory.h"
 #include "api/field_trials_view.h"
+#include "api/rtc_event_log/rtc_event.h"
 #include "api/rtc_event_log/rtc_event_log.h"
+#include "api/task_queue/task_queue_base.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "api/units/timestamp.h"
 #include "system_wrappers/include/clock.h"
--- a/third_party/libwebrtc/api/field_trials.cc
+++ b/third_party/libwebrtc/api/field_trials.cc
@ -11,8 +11,13 @@
 #include "api/field_trials.h"

 #include <atomic>
+#include <memory>
+#include <string>
+#include <utility>

+#include "absl/strings/string_view.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/containers/flat_map.h"
 #include "system_wrappers/include/field_trial.h"

 namespace {
--- a/third_party/libwebrtc/api/field_trials_registry.cc
+++ b/third_party/libwebrtc/api/field_trials_registry.cc
@ -11,12 +11,13 @@

 #include <string>

-#include "absl/algorithm/container.h"
 #include "absl/strings/string_view.h"
+// IWYU pragma: begin_keep
+#include "absl/algorithm/container.h"
 #include "experiments/registered_field_trials.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/containers/flat_set.h"
 #include "rtc_base/logging.h"
+// IWYU pragma: end_keep

 namespace webrtc {

--- a/third_party/libwebrtc/api/field_trials_unittest.cc
+++ b/third_party/libwebrtc/api/field_trials_unittest.cc
@ -11,9 +11,7 @@
 #include "api/field_trials.h"

 #include <memory>
-#include <utility>

-#include "absl/strings/string_view.h"
 #include "api/transport/field_trial_based_config.h"
 #include "rtc_base/containers/flat_set.h"
 #include "system_wrappers/include/field_trial.h"
--- a/third_party/libwebrtc/api/frame_transformer_factory.cc
+++ b/third_party/libwebrtc/api/frame_transformer_factory.cc
@ -10,8 +10,12 @@

 #include "api/frame_transformer_factory.h"

+#include <memory>
+
+#include "api/frame_transformer_interface.h"
 #include "audio/channel_send_frame_transformer_delegate.h"
 #include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"
+#include "rtc_base/checks.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/frame_transformer_factory.h
+++ b/third_party/libwebrtc/api/frame_transformer_factory.h
@ -12,12 +12,9 @@
 #define API_FRAME_TRANSFORMER_FACTORY_H_

 #include <memory>
-#include <vector>

 #include "api/frame_transformer_interface.h"
-#include "api/scoped_refptr.h"
-#include "api/video/encoded_frame.h"
-#include "api/video/video_frame_metadata.h"
+#include "rtc_base/system/rtc_export.h"

 // This file contains EXPERIMENTAL functions to create video frames from
 // either an old video frame or directly from parameters.
--- a/third_party/libwebrtc/api/function_view.h
+++ b/third_party/libwebrtc/api/function_view.h
@ -11,6 +11,7 @@
 #ifndef API_FUNCTION_VIEW_H_
 #define API_FUNCTION_VIEW_H_

+#include <cstddef>
 #include <type_traits>
 #include <utility>

--- a/third_party/libwebrtc/api/ice_transport_factory.cc
+++ b/third_party/libwebrtc/api/ice_transport_factory.cc
@ -13,12 +13,15 @@
 #include <memory>
 #include <utility>

+#include "api/ice_transport_interface.h"
 #include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
 #include "p2p/base/ice_transport_internal.h"
 #include "p2p/base/p2p_constants.h"
 #include "p2p/base/p2p_transport_channel.h"
 #include "p2p/base/port_allocator.h"
-#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/ice_transport_factory.h
+++ b/third_party/libwebrtc/api/ice_transport_factory.h
@ -16,7 +16,7 @@
 #include "rtc_base/system/rtc_export.h"

 namespace cricket {
-class PortAllocator;
+class PortAllocator;  // IWYU pragma: keep
 }  // namespace cricket

 namespace webrtc {
--- a/third_party/libwebrtc/api/legacy_stats_types.h
+++ b/third_party/libwebrtc/api/legacy_stats_types.h
@ -14,7 +14,9 @@
 #ifndef API_LEGACY_STATS_TYPES_H_
 #define API_LEGACY_STATS_TYPES_H_

-#include <algorithm>
+#include <stddef.h>
+#include <stdint.h>
+
 #include <list>
 #include <map>
 #include <string>
@ -23,7 +25,9 @@
 #include "api/ref_count.h"
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/system/rtc_export.h"
+#include "rtc_base/thread_annotations.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/media_stream_interface.h
+++ b/third_party/libwebrtc/api/media_stream_interface.h
@ -17,6 +17,7 @@
 #define API_MEDIA_STREAM_INTERFACE_H_

 #include <stddef.h>
+#include <stdint.h>

 #include <string>
 #include <vector>
@ -31,6 +32,7 @@
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
 #include "api/video_track_source_constraints.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/system/rtc_export.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/metronome/test/fake_metronome.cc
+++ b/third_party/libwebrtc/api/metronome/test/fake_metronome.cc
@ -10,6 +10,7 @@

 #include "api/metronome/test/fake_metronome.h"

+#include <cstddef>
 #include <utility>
 #include <vector>

--- a/third_party/libwebrtc/api/neteq/BUILD.gn
+++ b/third_party/libwebrtc/api/neteq/BUILD.gn
@ -17,6 +17,7 @@ rtc_source_set("neteq_api") {
  ]

  deps = [
+    "..:array_view",
    "..:rtp_headers",
    "..:rtp_packet_info",
    "..:scoped_refptr",
--- a/third_party/libwebrtc/api/neteq/custom_neteq_factory.h
+++ b/third_party/libwebrtc/api/neteq/custom_neteq_factory.h
@ -14,6 +14,7 @@
 #include <memory>

 #include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/neteq/neteq.h"
 #include "api/neteq/neteq_controller_factory.h"
 #include "api/neteq/neteq_factory.h"
 #include "api/scoped_refptr.h"
--- a/third_party/libwebrtc/api/neteq/default_neteq_controller_factory.h
+++ b/third_party/libwebrtc/api/neteq/default_neteq_controller_factory.h
@ -13,6 +13,7 @@

 #include <memory>

+#include "api/neteq/neteq_controller.h"
 #include "api/neteq/neteq_controller_factory.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/neteq/neteq.h
+++ b/third_party/libwebrtc/api/neteq/neteq.h
@ -12,25 +12,23 @@
 #define API_NETEQ_NETEQ_H_

 #include <stddef.h>  // Provide access to size_t.
+#include <stdint.h>

 #include <map>
 #include <string>
 #include <vector>

 #include "absl/types/optional.h"
+#include "api/array_view.h"
 #include "api/audio_codecs/audio_codec_pair_id.h"
-#include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/rtp_headers.h"
-#include "api/scoped_refptr.h"
 #include "api/units/timestamp.h"

 namespace webrtc {

 // Forward declarations.
 class AudioFrame;
-class AudioDecoderFactory;
-class Clock;

 struct NetEqNetworkStatistics {
  uint16_t current_buffer_size_ms;    // Current jitter buffer size in ms.
--- a/third_party/libwebrtc/api/neteq/tick_timer_unittest.cc
+++ b/third_party/libwebrtc/api/neteq/tick_timer_unittest.cc
@ -10,9 +10,9 @@

 #include "api/neteq/tick_timer.h"

+#include <cstdint>
 #include <memory>

-#include "test/gmock.h"
 #include "test/gtest.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/numerics/samples_stats_counter.h
+++ b/third_party/libwebrtc/api/numerics/samples_stats_counter.h
@ -11,6 +11,9 @@
 #ifndef API_NUMERICS_SAMPLES_STATS_COUNTER_H_
 #define API_NUMERICS_SAMPLES_STATS_COUNTER_H_

+#include <stddef.h>
+#include <stdint.h>
+
 #include <map>
 #include <string>
 #include <vector>
--- a/third_party/libwebrtc/api/peer_connection_interface.h
+++ b/third_party/libwebrtc/api/peer_connection_interface.h
@ -66,6 +66,7 @@

 #ifndef API_PEER_CONNECTION_INTERFACE_H_
 #define API_PEER_CONNECTION_INTERFACE_H_
+// IWYU pragma: no_include "pc/media_factory.h"

 #include <stdint.h>
 #include <stdio.h>
@ -80,7 +81,9 @@
 #include "absl/types/optional.h"
 #include "api/adaptation/resource.h"
 #include "api/async_dns_resolver.h"
+#include "api/audio/audio_device.h"
 #include "api/audio/audio_mixer.h"
+#include "api/audio/audio_processing.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_options.h"
@ -123,32 +126,36 @@
 #include "api/video_codecs/video_encoder_factory.h"
 #include "call/rtp_transport_controller_send_factory_interface.h"
 #include "media/base/media_config.h"
-#include "media/base/media_engine.h"
 // TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
 // inject a PacketSocketFactory and/or NetworkManager, and not expose
 // PortAllocator in the PeerConnection api.
+#include "api/audio/audio_frame_processor.h"
 #include "api/ref_count.h"
+#include "api/units/time_delta.h"
+#include "p2p/base/port.h"
 #include "p2p/base/port_allocator.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/network.h"
 #include "rtc_base/network_constants.h"
 #include "rtc_base/network_monitor_factory.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/rtc_certificate_generator.h"
-#include "rtc_base/socket_address.h"
+#include "rtc_base/socket_factory.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/system/rtc_export.h"
 #include "rtc_base/thread.h"

 namespace rtc {
-class Thread;
+class Thread;  // IWYU pragma: keep
 }  // namespace rtc

 namespace webrtc {
-
+// IWYU pragma: begin_keep
 // MediaFactory class definition is not part of the api.
 class MediaFactory;

+// IWYU pragma: end_keep
 // MediaStream container interface.
 class StreamCollectionInterface : public webrtc::RefCountInterface {
 public:
--- a/third_party/libwebrtc/api/rtc_error.h
+++ b/third_party/libwebrtc/api/rtc_error.h
@ -14,6 +14,8 @@
 #ifdef WEBRTC_UNIT_TEST
 #include <ostream>
 #endif  // WEBRTC_UNIT_TEST
+#include <stdint.h>
+
 #include <string>
 #include <utility>  // For std::move.

--- a/third_party/libwebrtc/api/rtp_headers.h
+++ b/third_party/libwebrtc/api/rtp_headers.h
@ -17,12 +17,13 @@
 #include <string>

 #include "absl/types/optional.h"
-#include "api/array_view.h"
 #include "api/units/timestamp.h"
 #include "api/video/color_space.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/rtc_export.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/rtp_packet_info.cc
+++ b/third_party/libwebrtc/api/rtp_packet_info.cc
@ -10,8 +10,15 @@

 #include "api/rtp_packet_info.h"

+#include <stddef.h>
+
 #include <algorithm>
+#include <cstdint>
 #include <utility>
+#include <vector>
+
+#include "api/rtp_headers.h"
+#include "api/units/timestamp.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/rtp_packet_info_unittest.cc
+++ b/third_party/libwebrtc/api/rtp_packet_info_unittest.cc
@ -8,9 +8,15 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "api/rtp_packet_infos.h"
+#include "api/rtp_packet_info.h"
+
+#include <cstdint>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/rtp_headers.h"
 #include "api/units/time_delta.h"
-#include "test/gmock.h"
+#include "api/units/timestamp.h"
 #include "test/gtest.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/rtp_packet_infos.h
+++ b/third_party/libwebrtc/api/rtp_packet_infos.h
@ -11,7 +11,6 @@
 #ifndef API_RTP_PACKET_INFOS_H_
 #define API_RTP_PACKET_INFOS_H_

-#include <cstdint>
 #include <utility>
 #include <vector>

--- a/third_party/libwebrtc/api/rtp_packet_infos_unittest.cc
+++ b/third_party/libwebrtc/api/rtp_packet_infos_unittest.cc
@ -10,6 +10,11 @@

 #include "api/rtp_packet_infos.h"

+#include <stddef.h>
+
+#include "api/rtp_headers.h"
+#include "api/rtp_packet_info.h"
+#include "api/units/timestamp.h"
 #include "test/gmock.h"
 #include "test/gtest.h"

--- a/third_party/libwebrtc/api/rtp_parameters.cc
+++ b/third_party/libwebrtc/api/rtp_parameters.cc
@ -10,12 +10,16 @@
 #include "api/rtp_parameters.h"

 #include <algorithm>
+#include <cstdint>
 #include <string>
 #include <tuple>
-#include <utility>
+#include <vector>

+#include "absl/strings/string_view.h"
 #include "api/array_view.h"
+#include "api/rtp_transceiver_direction.h"
 #include "media/base/media_constants.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/strings/string_builder.h"

 namespace webrtc {
--- a/third_party/libwebrtc/api/rtp_transceiver_interface.h
+++ b/third_party/libwebrtc/api/rtp_transceiver_interface.h
@ -19,6 +19,7 @@
 #include "api/array_view.h"
 #include "api/media_types.h"
 #include "api/ref_count.h"
+#include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
 #include "api/rtp_receiver_interface.h"
 #include "api/rtp_sender_interface.h"
--- a/third_party/libwebrtc/api/scoped_refptr.h
+++ b/third_party/libwebrtc/api/scoped_refptr.h
@ -63,7 +63,7 @@
 #ifndef API_SCOPED_REFPTR_H_
 #define API_SCOPED_REFPTR_H_

-#include <memory>
+#include <cstddef>
 #include <utility>

 namespace webrtc {
--- a/third_party/libwebrtc/api/scoped_refptr_unittest.cc
+++ b/third_party/libwebrtc/api/scoped_refptr_unittest.cc
@ -10,6 +10,7 @@

 #include "api/scoped_refptr.h"

+#include <type_traits>
 #include <utility>
 #include <vector>

--- a/third_party/libwebrtc/api/sctp_transport_interface.h
+++ b/third_party/libwebrtc/api/sctp_transport_interface.h
@ -14,8 +14,8 @@
 #include "absl/types/optional.h"
 #include "api/dtls_transport_interface.h"
 #include "api/ref_count.h"
-#include "api/rtc_error.h"
 #include "api/scoped_refptr.h"
+#include "rtc_base/system/rtc_export.h"

 namespace webrtc {

--- a/third_party/libwebrtc/api/sequence_checker.h
+++ b/third_party/libwebrtc/api/sequence_checker.h
@ -10,6 +10,7 @@
 #ifndef API_SEQUENCE_CHECKER_H_
 #define API_SEQUENCE_CHECKER_H_

+#include "api/task_queue/task_queue_base.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/synchronization/sequence_checker_internal.h"
 #include "rtc_base/thread_annotations.h"
--- a/third_party/libwebrtc/api/sequence_checker_unittest.cc
+++ b/third_party/libwebrtc/api/sequence_checker_unittest.cc
@ -10,14 +10,18 @@

 #include "api/sequence_checker.h"

+#include <functional>
 #include <memory>
-#include <utility>

+#include "absl/functional/any_invocable.h"
 #include "api/function_view.h"
 #include "api/units/time_delta.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/sequence_checker_internal.h"
 #include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/thread_annotations.h"
 #include "test/gmock.h"
 #include "test/gtest.h"

--- a/third_party/libwebrtc/api/voip/test/compile_all_headers.cc
+++ b/third_party/libwebrtc/api/voip/test/compile_all_headers.cc
@ -11,4 +11,3 @@
 // This file verifies that all include files in this directory can be
 // compiled without errors or other required includes.

-#include "api/voip/test/mock_voip_engine.h"
--- a/third_party/libwebrtc/modules/audio_coding/BUILD.gn
+++ b/third_party/libwebrtc/modules/audio_coding/BUILD.gn
@ -505,6 +505,7 @@ rtc_library("webrtc_multiopus") {

  deps = [
    ":audio_coding_opus_common",
+    "../../api:array_view",
    "../../api/audio_codecs:audio_codecs_api",
    "../../api/audio_codecs/opus:audio_decoder_opus_config",
    "../../api/audio_codecs/opus:audio_encoder_opus_config",
@ -513,6 +514,7 @@ rtc_library("webrtc_multiopus") {
    "../../rtc_base:checks",
    "../../rtc_base:logging",
    "../../rtc_base:macromagic",
+    "../../rtc_base:safe_conversions",
    "../../rtc_base:safe_minmax",
    "../../rtc_base:stringutils",
    "//third_party/abseil-cpp/absl/memory",
--- a/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.h
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.h
@ -13,6 +13,7 @@

 #include <vector>

+#include "api/array_view.h"
 #include "api/audio/audio_frame.h"

 namespace webrtc {
--- a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.cc
@ -20,15 +20,17 @@
 #include "modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h"

 #include <algorithm>
+#include <iterator>
 #include <memory>
 #include <string>
 #include <vector>

 #include "absl/strings/match.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
 #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
-#include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/string_to_number.h"

 namespace webrtc {
--- a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_multi_channel_opus_impl.h
@ -11,16 +11,21 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_
 #define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_MULTI_CHANNEL_OPUS_IMPL_H_

+#include <stddef.h>
+#include <stdint.h>
+
 #include <memory>
 #include <utility>
 #include <vector>

 #include "absl/types/optional.h"
+#include "api/array_view.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/audio_codecs/opus/audio_encoder_multi_channel_opus_config.h"
 #include "api/units/time_delta.h"
 #include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/buffer.h"

 namespace webrtc {

--- a/third_party/libwebrtc/moz-patch-stack/0002.patch
+++ b/third_party/libwebrtc/moz-patch-stack/0002.patch
@ -363,10 +363,10 @@ index 8813e0f4c2..b5e08cea2c 100644
 RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
     default;
 diff --git a/api/rtp_headers.h b/api/rtp_headers.h
-index 7ededb94fc..bf1660d934 100644
+index 129ab5f499..5a867bbd77 100644
 --- a/api/rtp_headers.h
 +++ b/api/rtp_headers.h
-@@ -112,6 +112,19 @@ inline bool operator!=(const AbsoluteCaptureTime& lhs,
+@@ -113,6 +113,19 @@ inline bool operator!=(const AbsoluteCaptureTime& lhs,
   return !(lhs == rhs);
 }
 
@ -386,7 +386,7 @@ index 7ededb94fc..bf1660d934 100644
 struct RTPHeaderExtension {
   RTPHeaderExtension();
   RTPHeaderExtension(const RTPHeaderExtension& other);
-@@ -170,12 +183,12 @@ struct RTPHeaderExtension {
+@@ -171,12 +184,12 @@ struct RTPHeaderExtension {
 
   absl::optional<ColorSpace> color_space;
 
@ -402,10 +402,10 @@ index 7ededb94fc..bf1660d934 100644
   RTPHeader();
   RTPHeader(const RTPHeader& other);
 diff --git a/api/rtp_parameters.cc b/api/rtp_parameters.cc
-index 3ff4b58a2e..ad0f3c9396 100644
+index 283d238e77..da06138ce8 100644
 --- a/api/rtp_parameters.cc
 +++ b/api/rtp_parameters.cc
-@@ -160,7 +160,8 @@ bool RtpExtension::IsSupportedForAudio(absl::string_view uri) {
+@@ -164,7 +164,8 @@ bool RtpExtension::IsSupportedForAudio(absl::string_view uri) {
          uri == webrtc::RtpExtension::kTransportSequenceNumberV2Uri ||
          uri == webrtc::RtpExtension::kMidUri ||
          uri == webrtc::RtpExtension::kRidUri ||
--- a/third_party/libwebrtc/moz-patch-stack/0031.patch
+++ b/third_party/libwebrtc/moz-patch-stack/0031.patch
@ -206,10 +206,10 @@ index 7328df134a..c8bda37ad6 100644
     } else {
       deps += [
 diff --git a/api/BUILD.gn b/api/BUILD.gn
-index c68e319da7..0907ed1c88 100644
+index e537068ab5..2710b2476e 100644
 --- a/api/BUILD.gn
 +++ b/api/BUILD.gn
-@@ -40,6 +40,9 @@ rtc_source_set("enable_media") {
+@@ -44,6 +44,9 @@ rtc_source_set("enable_media") {
     "../rtc_base/system:rtc_export",
     "environment",
   ]
@ -219,7 +219,7 @@ index c68e319da7..0907ed1c88 100644
 }
 
 rtc_source_set("enable_media_with_defaults") {
-@@ -66,7 +69,7 @@ rtc_source_set("enable_media_with_defaults") {
+@@ -76,7 +79,7 @@ rtc_source_set("enable_media_with_defaults") {
   ]
 }
 
@ -228,7 +228,7 @@ index c68e319da7..0907ed1c88 100644
   rtc_library("create_peerconnection_factory") {
     visibility = [ "*" ]
     allow_poison = [ "environment_construction" ]
-@@ -217,6 +220,10 @@ rtc_source_set("ice_transport_interface") {
+@@ -230,6 +233,10 @@ rtc_source_set("ice_transport_interface") {
 }
 
 rtc_library("dtls_transport_interface") {
@ -239,7 +239,7 @@ index c68e319da7..0907ed1c88 100644
   visibility = [ "*" ]
 
   sources = [
-@@ -233,6 +240,7 @@ rtc_library("dtls_transport_interface") {
+@@ -247,6 +254,7 @@ rtc_library("dtls_transport_interface") {
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@ -247,7 +247,7 @@ index c68e319da7..0907ed1c88 100644
 
 rtc_library("dtmf_sender_interface") {
   visibility = [ "*" ]
-@@ -245,6 +253,10 @@ rtc_library("dtmf_sender_interface") {
+@@ -259,6 +267,10 @@ rtc_library("dtmf_sender_interface") {
 }
 
 rtc_library("rtp_sender_interface") {
@ -258,7 +258,7 @@ index c68e319da7..0907ed1c88 100644
   visibility = [ "*" ]
 
   sources = [
-@@ -259,6 +271,7 @@ rtc_library("rtp_sender_interface") {
+@@ -273,6 +285,7 @@ rtc_library("rtp_sender_interface") {
     ":ref_count",
     ":rtc_error",
     ":rtp_parameters",
@ -266,7 +266,7 @@ index c68e319da7..0907ed1c88 100644
     ":scoped_refptr",
     "../rtc_base:checks",
     "../rtc_base/system:rtc_export",
-@@ -267,8 +280,23 @@ rtc_library("rtp_sender_interface") {
+@@ -281,8 +294,23 @@ rtc_library("rtp_sender_interface") {
     "//third_party/abseil-cpp/absl/functional:any_invocable",
   ]
 }
@ -290,7 +290,7 @@ index c68e319da7..0907ed1c88 100644
   visibility = [ "*" ]
   cflags = []
   sources = [
-@@ -385,6 +413,7 @@ rtc_library("libjingle_peerconnection_api") {
+@@ -405,6 +433,7 @@ rtc_library("libjingle_peerconnection_api") {
     "../rtc_base/system:rtc_export",
   ]
 }
@ -298,7 +298,7 @@ index c68e319da7..0907ed1c88 100644
 
 rtc_source_set("frame_transformer_interface") {
   visibility = [ "*" ]
-@@ -560,6 +589,7 @@ rtc_source_set("peer_network_dependencies") {
+@@ -580,6 +609,7 @@ rtc_source_set("peer_network_dependencies") {
 }
 
 rtc_source_set("peer_connection_quality_test_fixture_api") {
@ -306,7 +306,7 @@ index c68e319da7..0907ed1c88 100644
   visibility = [ "*" ]
   testonly = true
   sources = [ "test/peerconnection_quality_test_fixture.h" ]
-@@ -606,6 +636,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
+@@ -626,6 +656,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@ -314,7 +314,7 @@ index c68e319da7..0907ed1c88 100644
 
 rtc_source_set("frame_generator_api") {
   visibility = [ "*" ]
-@@ -729,6 +760,7 @@ rtc_library("create_frame_generator") {
+@@ -749,6 +780,7 @@ rtc_library("create_frame_generator") {
   ]
 }
 
@ -322,7 +322,7 @@ index c68e319da7..0907ed1c88 100644
 rtc_library("create_peer_connection_quality_test_frame_generator") {
   visibility = [ "*" ]
   testonly = true
-@@ -745,6 +777,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
+@@ -765,6 +797,7 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@ -330,7 +330,7 @@ index c68e319da7..0907ed1c88 100644
 
 rtc_source_set("libjingle_logging_api") {
   visibility = [ "*" ]
-@@ -925,6 +958,7 @@ rtc_source_set("refcountedbase") {
+@@ -945,6 +978,7 @@ rtc_source_set("refcountedbase") {
   ]
 }
 
@ -338,7 +338,7 @@ index c68e319da7..0907ed1c88 100644
 rtc_library("ice_transport_factory") {
   visibility = [ "*" ]
   sources = [
-@@ -948,6 +982,7 @@ rtc_library("ice_transport_factory") {
+@@ -970,6 +1004,7 @@ rtc_library("ice_transport_factory") {
     "rtc_event_log:rtc_event_log",
   ]
 }
@ -760,10 +760,10 @@ index 1c08382969..ff69ea62dc 100644
 using webrtc::FrameDecryptorInterface;
 using webrtc::FrameEncryptorInterface;
 diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
-index 655a9347fe..2ba849ff68 100644
+index 5416f83901..316b3db72c 100644
 --- a/modules/audio_coding/BUILD.gn
 +++ b/modules/audio_coding/BUILD.gn
-@@ -546,7 +546,7 @@ rtc_library("webrtc_opus_wrapper") {
+@@ -548,7 +548,7 @@ rtc_library("webrtc_opus_wrapper") {
     deps += [ rtc_opus_dir ]
     public_configs = [ "//third_party/opus:opus_config" ]
   } else if (build_with_mozilla) {
--- a/third_party/libwebrtc/moz-patch-stack/0066.patch
+++ b/third_party/libwebrtc/moz-patch-stack/0066.patch
@ -26,7 +26,7 @@ index 7ac93125f8..6003780e37 100644
       deps += [ "logging:rtc_event_log_proto" ]
     }
 diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
-index e7ec3e3614..ff4b6d1deb 100644
+index 380d2dce62..f7a50fa932 100644
 --- a/sdk/BUILD.gn
 +++ b/sdk/BUILD.gn
@@ -528,6 +528,7 @@ if (is_ios || is_mac) {
@ -69,7 +69,7 @@ index e7ec3e3614..ff4b6d1deb 100644
     rtc_library("videocodec_objc") {
       visibility = [ "*" ]
       configs += [ "..:no_global_constructors" ]
-@@ -1750,5 +1755,6 @@ if (is_ios || is_mac) {
+@@ -1754,5 +1759,6 @@ if (is_ios || is_mac) {
         "VideoToolbox.framework",
       ]
     }
--- a/third_party/libwebrtc/moz-patch-stack/0102.patch
+++ b/third_party/libwebrtc/moz-patch-stack/0102.patch
@ -23,10 +23,10 @@ index 6003780e37..3c8a589c04 100644
         "api/video:video_rtp_headers",
         "test:rtp_test_utils",
 diff --git a/api/environment/environment_factory.cc b/api/environment/environment_factory.cc
-index c0b681aa08..6f0ec40dbe 100644
+index 293ef6cf8a..b9cb6b4f53 100644
 --- a/api/environment/environment_factory.cc
 +++ b/api/environment/environment_factory.cc
-@@ -97,12 +97,22 @@ Environment EnvironmentFactory::CreateWithDefaults() && {
+@@ -102,12 +102,22 @@ Environment EnvironmentFactory::CreateWithDefaults() && {
   if (field_trials_ == nullptr) {
     Set(std::make_unique<FieldTrialBasedConfig>());
   }
--- a/third_party/libwebrtc/rtc_base/synchronization/BUILD.gn
+++ b/third_party/libwebrtc/rtc_base/synchronization/BUILD.gn
@ -44,7 +44,10 @@ rtc_source_set("mutex") {
 }

 rtc_library("sequence_checker_internal") {
-  visibility = [ "../../api:sequence_checker" ]
+  visibility = [
+    "../../api:rtc_api_unittests",
+    "../../api:sequence_checker",
+  ]
  sources = [
    "sequence_checker_internal.cc",
    "sequence_checker_internal.h",
--- a/third_party/libwebrtc/rtc_tools/data_channel_benchmark/BUILD.gn
+++ b/third_party/libwebrtc/rtc_tools/data_channel_benchmark/BUILD.gn
@ -46,8 +46,13 @@ rtc_executable("data_channel_benchmark") {
    "../../api:libjingle_peerconnection_api",
    "../../api:rtc_error",
    "../../api:scoped_refptr",
+    "../../api/audio:audio_device",
+    "../../api/audio:audio_mixer_api",
+    "../../api/audio:audio_processing",
    "../../api/audio_codecs:builtin_audio_decoder_factory",
    "../../api/audio_codecs:builtin_audio_encoder_factory",
+    "../../api/video_codecs:video_codecs_api",
+    "../../api/video_codecs:video_codecs_api",
    "../../api/video_codecs:video_decoder_factory_template",
    "../../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
    "../../api/video_codecs:video_decoder_factory_template_libvpx_vp8_adapter",
@ -58,6 +63,7 @@ rtc_executable("data_channel_benchmark") {
    "../../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
    "../../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
    "../../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
+    "../../rtc_base:checks",
    "../../rtc_base:logging",
    "../../rtc_base:refcount",
    "../../rtc_base:rtc_event",
--- a/third_party/libwebrtc/rtc_tools/data_channel_benchmark/peer_connection_client.cc
+++ b/third_party/libwebrtc/rtc_tools/data_channel_benchmark/peer_connection_client.cc
@ -13,6 +13,9 @@
 #include <string>
 #include <utility>

+#include "api/audio/audio_device.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio/audio_processing.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "api/create_peerconnection_factory.h"
@ -20,18 +23,23 @@
 #include "api/peer_connection_interface.h"
 #include "api/rtc_error.h"
 #include "api/scoped_refptr.h"
+#include "api/set_local_description_observer_interface.h"
 #include "api/set_remote_description_observer_interface.h"
+#include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_decoder_factory_template.h"
 #include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
 #include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h"
 #include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h"
 #include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h"
+#include "api/video_codecs/video_encoder_factory.h"
 #include "api/video_codecs/video_encoder_factory_template.h"
 #include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
 #include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/thread.h"

 namespace {
--- a/third_party/libwebrtc/rtc_tools/data_channel_benchmark/peer_connection_client.h
+++ b/third_party/libwebrtc/rtc_tools/data_channel_benchmark/peer_connection_client.h
@ -12,10 +12,12 @@

 #include <stdint.h>

+#include <functional>
 #include <memory>
 #include <string>
 #include <vector>

+#include "api/data_channel_interface.h"
 #include "api/jsep.h"
 #include "api/peer_connection_interface.h"
 #include "api/rtp_receiver_interface.h"
--- a/third_party/libwebrtc/sdk/BUILD.gn
+++ b/third_party/libwebrtc/sdk/BUILD.gn
@ -1079,6 +1079,7 @@ if (is_ios || is_mac) {
        "../api/rtc_event_log:rtc_event_log_factory",
        "../api/task_queue:default_task_queue_factory",
        "../api/transport:field_trial_based_config",
+        "../api/transport:network_control",
        "../api/transport/rtp:rtp_source",
        "../api/video:video_frame",
        "../api/video:video_rtp_headers",
@ -1107,7 +1108,10 @@ if (is_ios || is_mac) {
      ]

      if (is_ios) {
-        deps += [ ":native_api_audio_device_module" ]
+        deps += [
+          ":native_api_audio_device_module",
+          "../api/transport:network_control",
+        ]
      }
    }

--- a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
@ -10,17 +10,17 @@

 #import "RTCPeerConnectionFactory.h"

+#include "api/audio/audio_device.h"
+#include "api/audio/audio_processing.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/scoped_refptr.h"
+#include "api/transport/network_control.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"

 namespace webrtc {

-class AudioDeviceModule;
-class AudioEncoderFactory;
-class AudioDecoderFactory;
-class NetworkControllerFactoryInterface;
-class VideoEncoderFactory;
-class VideoDecoderFactory;
-class AudioProcessing;
 struct PeerConnectionDependencies;

 }  // namespace webrtc
--- a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.h
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.h
@ -10,18 +10,13 @@

 #import "RTCPeerConnectionFactory.h"

+#include "api/audio/audio_device.h"
+#include "api/audio/audio_processing.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/scoped_refptr.h"
-
-namespace webrtc {
-
-class AudioDeviceModule;
-class AudioEncoderFactory;
-class AudioDecoderFactory;
-class VideoEncoderFactory;
-class VideoDecoderFactory;
-class AudioProcessing;
-
-}  // namespace webrtc
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"

 NS_ASSUME_NONNULL_BEGIN

--- a/third_party/libwebrtc/test/pc/e2e/BUILD.gn
+++ b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
@ -164,12 +164,18 @@ if (!build_with_chromium) {
      ]
      deps = [
        "../..:fileutils",
+        "../../../api:array_view",
        "../../../api:peer_network_dependencies",
+        "../../../api:rtp_parameters",
        "../../../api/test/pclf:media_configuration",
        "../../../api/test/pclf:media_quality_test_params",
        "../../../api/test/pclf:peer_configurer",
+        "../../../api/video_codecs:scalability_mode",
+        "../../../media:media_constants",
        "../../../modules/video_coding/svc:scalability_mode_util",
        "../../../modules/video_coding/svc:scalability_structures",
+        "../../../modules/video_coding/svc:scalable_video_controller",
+        "../../../rtc_base:checks",
        "../../../rtc_base:macromagic",
        "//third_party/abseil-cpp/absl/strings:string_view",
      ]
--- a/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.cc
+++ b/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.cc
@ -10,17 +10,26 @@

 #include "test/pc/e2e/peer_params_preprocessor.h"

+#include <stddef.h>
+
+#include <optional>
 #include <set>
 #include <string>
+#include <utility>
+#include <vector>

 #include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/rtp_parameters.h"
 #include "api/test/pclf/media_configuration.h"
 #include "api/test/pclf/media_quality_test_params.h"
 #include "api/test/pclf/peer_configurer.h"
-#include "api/test/peer_network_dependencies.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "media/base/media_constants.h"
 #include "modules/video_coding/svc/create_scalability_structure.h"
 #include "modules/video_coding/svc/scalability_mode_util.h"
-#include "rtc_base/arraysize.h"
+#include "modules/video_coding/svc/scalable_video_controller.h"
+#include "rtc_base/checks.h"
 #include "test/testsupport/file_utils.h"

 namespace webrtc {